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| author | Max Kellermann <max@duempel.org> | 2012-03-27 00:22:47 +0200 |
|---|---|---|
| committer | Max Kellermann <max@duempel.org> | 2012-03-27 01:22:50 +0200 |
| commit | d271dd2cce79d1473f43f7ad7e6815b74617dfe3 (patch) | |
| tree | 9ae91e3a33ba41640ad25b1974278d766ef2ef1f /src/output | |
| parent | ebfdd37451cea0151de74005d83db5ac31fcaf77 (diff) | |
| download | mpd-d271dd2cce79d1473f43f7ad7e6815b74617dfe3.tar.gz mpd-d271dd2cce79d1473f43f7ad7e6815b74617dfe3.tar.xz mpd-d271dd2cce79d1473f43f7ad7e6815b74617dfe3.zip | |
output/alsa: use pcm_export for the DSD-over-USB conversion
Diffstat (limited to 'src/output')
| -rw-r--r-- | src/output/alsa_output_plugin.c | 21 |
1 files changed, 10 insertions, 11 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c index d131003e3..00ddbfb5c 100644 --- a/src/output/alsa_output_plugin.c +++ b/src/output/alsa_output_plugin.c @@ -595,15 +595,16 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format, /* pass 24 bit to alsa_setup() */ - audio_format->format = SAMPLE_FORMAT_S24_P32; - audio_format->sample_rate /= 2; + struct audio_format usb_format = *audio_format; + usb_format.format = SAMPLE_FORMAT_S24_P32; + usb_format.sample_rate /= 2; - const struct audio_format check = *audio_format; + const struct audio_format check = usb_format; - if (!alsa_setup(ad, audio_format, packed_r, reverse_endian_r, error_r)) + if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r)) return false; - if (!audio_format_equals(audio_format, &check)) { + if (!audio_format_equals(&usb_format, &check)) { /* no bit-perfect playback, which is required for DSD over USB */ g_set_error(error_r, alsa_output_quark(), 0, @@ -612,10 +613,6 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format, return false; } - /* the ALSA device has accepted 24 bit playback, - return DSD_OVER_USB to the caller */ - - audio_format->format = SAMPLE_FORMAT_DSD_OVER_USB; return true; } @@ -625,7 +622,9 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format, { bool packed, reverse_endian; - const bool success = ad->dsd_usb && audio_format->format == SAMPLE_FORMAT_DSD + const bool dsd_usb = ad->dsd_usb && + audio_format->format == SAMPLE_FORMAT_DSD; + const bool success = dsd_usb ? alsa_setup_dsd(ad, audio_format, &packed, &reverse_endian, error_r) : alsa_setup(ad, audio_format, &packed, &reverse_endian, @@ -635,7 +634,7 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format, pcm_export_open(&ad->export, audio_format->format, audio_format->channels, - false, packed, reverse_endian); + dsd_usb, packed, reverse_endian); return true; } |